Summary:ASTERISK-16412: [patch] [regression]context value from chan_dahdi.conf not used.
Reporter:iasgoscouk (iasgoscouk)Labels:
Date Opened:2010-07-24 04:45:07Date Closed:2010-07-26 22:02:33
Versions:Frequency of
Environment:Attachments:( 0) chan_dahdi.conf
( 1) extensions.conf
( 2) issue17693.patch
( 3) mygosLog_1.6.2_11_rc1.txt
( 4) mygosLog_1.8_beta1_patched.txt
( 5) mygosLog_1.8_beta1.txt
Description:Am currently running ok with, and this morning tried 1.8 beta.

When receiving incoming POTS call via DAHDI, the following error occurs:

[Jul 24 10:19:31] WARNING[2065] pbx.c: Channel 'DAHDI/1-1' sent into invalid extension 's' in context 'default', but no invalid handler


I have seen a anumber of similar reported threads searching via google;   have checked the latest config's to see if anything has changed.  

Just reverted back to and everything is working ok again.
Comments:By: iasgoscouk (iasgoscouk) 2010-07-24 04:49:02

Not sure what specific details are required.    I have to revert back to the previous version to have a working version.  

I can easily revert back to 1.8 beta to give additional information.

By: Paul Belanger (pabelanger) 2010-07-24 07:26:18

Have you read UPGRADE.txt and CHANGES files?  Don't expect 1.6.2 configs to work with 1.8.  Aside from that, we will need a debug log (see below) and relevant .config files.

We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue:


By: iasgoscouk (iasgoscouk) 2010-07-24 07:44:11

I am a professional applications manager and  web developer; I have read all the configs as I said.  I have used UltraCompare to compare my configs to the samples, are checked through any differences and make them similar where appropriate.

I would have thought that such a major rewrite, if the case that the 1.6 configs won't work, that this would be documented properly in a 1.6 to 1.8 documents, rather than just a list of changes.   I have read throught both CHANGES and UPGRADE.

I don't normally bother reporting issues;  have worked with asterisk for many years and never bothered before.

For this upgrade I cannot use my Vmware installations for Fedora, so will have to wait for a few hours, until i can take down my production server again to produce the debug info as specified.

By: Paul Belanger (pabelanger) 2010-07-24 08:09:13

The problem clearly states, invalid extension 's' in context 'default'. So, this looks to be a configuration problem.  Attach your extensions.conf to the issue, or use:

*CLI> dialpan show s@default

I'd still want to see a debug log from a work (1.6.2) and non-working (1.8) system.  It is possible this is a regressions, but will have to wait for the logs first.

By: iasgoscouk (iasgoscouk) 2010-07-24 08:14:46

my 's' extension does not exist in [default].     my chan_dahdi.conf specifies a different context for pstn incoming calls.

when i originally checked the chan_dahdi against the sample, the layout of the file contains a vastly different layout.  

I didn't see anything anywhere to specify the that 's' extension should be in [default].

my chan-dahdi.conf is a cut-down version without comments etc to mae it more manageable.

i have attached my chan_dahdi.conf and extensions.conf

By: iasgoscouk (iasgoscouk) 2010-07-24 08:21:00

I am not able to do the 1.6.2 and 1.8 debug logs until about 18:00 BST, as the server is in use during daytime hours for web service/dns and other applications, and obviously my incoming/outgoing daily BT calls.

By: Paul Belanger (pabelanger) 2010-07-24 08:26:49

We'll also need the output of:

*CLI> dahdi show channel 1  ; or the actually channel number having the problem.  

It looks like the context value in chan_dahdi.conf is being ignored, why?  Thats the question.

By: iasgoscouk (iasgoscouk) 2010-07-24 08:38:03

enclosed file for current installtion whilst i knew about a due incoming call.  

there is a 'dahdi show channel 1' just prior to the incoming call.

i will hope to test against 1.8 in a short while.

By: iasgoscouk (iasgoscouk) 2010-07-24 11:58:42

enclosed file for 1.8 beta 1.

I removed all asterisk modules and  ran a full make clean;  ./configure;   make  etc again.

I noticed that a 'service asterisk start' following the build, on logging into asterisk console, it disconnected, within a few seconds, and had to log back in again.  

The attached debug logs include SIP registrations for my 2 sip providers too.

By: iasgoscouk (iasgoscouk) 2010-07-24 12:00:56

will not be able to pick up any updates from you now until 07:00 BST tomorrow (Sunday 25th)


By: Paul Belanger (pabelanger) 2010-07-24 14:17:18

Ok, I see the problem.  Let me see if I can write a patch.

By: iasgoscouk (iasgoscouk) 2010-07-24 14:20:51

ok thanks  =   i am online.    i have reverted to 1.6.2   =   if this is something with 1.8 will be happy to wait for a newer release, as long as this is something which helps the release of 1.8

By: Paul Belanger (pabelanger) 2010-07-24 15:40:31

Ok, give this patch a go.  It _should_ fall back to context if dringXcontext is empty.

Edit: It also includes some minor formatting changes.

By: iasgoscouk (iasgoscouk) 2010-07-25 01:31:32

Thanks - patch downloaded and applied ok.

Will feedback as soon as I have re-made and tested.

By: iasgoscouk (iasgoscouk) 2010-07-25 03:05:51

Test complete.   Incoming calls now working ok.

Enclosed copy of debug log.   This appears to be resolved - Thank you.

I will leave this version running.

Additional Comment of further testing of 1.8
I occasionally look at the SVN for asterisk, and a few weeks ago, there was an issue with mixmonitor incoming channel being out of sync.    I have checked the recordings on the 1.8 beta, and are they are all OK too.

By: Paul Belanger (pabelanger) 2010-07-25 07:55:02

Promoted to 'Ready for Review'

By: Digium Subversion (svnbot) 2010-07-26 21:57:32

Repository: asterisk
Revision: 279755

U   branches/1.8/channels/chan_dahdi.c

r279755 | pabelanger | 2010-07-26 21:57:31 -0500 (Mon, 26 Jul 2010) | 10 lines

If dringXcontext is null, fallback to default context value.

(closes issue ASTERISK-16412)
Reported by: iasgoscouk
     issue17693.patch uploaded by pabelanger (license 224)
Tested by: iasgoscouk

Review: https://reviewboard.asterisk.org/r/803/



By: Digium Subversion (svnbot) 2010-07-26 22:02:32

Repository: asterisk
Revision: 279756

_U  trunk/
U   trunk/channels/chan_dahdi.c

r279756 | pabelanger | 2010-07-26 22:02:32 -0500 (Mon, 26 Jul 2010) | 17 lines

Merged revisions 279755 via svnmerge from

 r279755 | pabelanger | 2010-07-26 22:57:33 -0400 (Mon, 26 Jul 2010) | 10 lines
 If dringXcontext is null, fallback to default context value.
 (closes issue ASTERISK-16412)
 Reported by: iasgoscouk
       issue17693.patch uploaded by pabelanger (license 224)
 Tested by: iasgoscouk
 Review: https://reviewboard.asterisk.org/r/803/